Hi, You need to launch the application twice! once on each machine. It stops at the end of your log because it is waiting for a connection, so if you don't launch a second test-rtp-3 from a second machine, then nothing will work.
Also, you specify port 8548, that doesn't look right, because as Olivier told you, the test listens on port 7078 : (lt-test-rtp-3:6724): farsight-transmitter-DEBUG: bound to port 7078 So you would need to launch it on both pcs, specifying the ip of the other pc, and specifying port 7078 on each sides. Hope that helps, KaKaRoTo souilem fadi wrote: > Hi everybody , thank's for the advice and specially to Mr .olivier . > i've compiled the example test-rtp-3 , and finnaly i'm capturing > packets with Tcpdump . but nothing happens anymore , i mean that the > test stop's in a strange manner . i'm sending you the log of the > execution hopping to receive an ansewer about how to use this code to > send or receive a stream . > > > // > ./test-rtp-3 192.168.1.114 8548 > > farsight-Message: looking for plugins in > /home/g545952/Desktop/fado//lib/farsight-0.1-2 > opening module > /home/g545952/Desktop/fado//lib/farsight-0.1-2/librtp-session.so: > succeeded > > protocol details: > name: rtp-session > description: Farsight RTP plugin > author: Farsight Project > > farsight-Message: looking for plugins in > /home/g545952/Desktop/fado//lib/farsight-0.1-2 > opening module > /home/g545952/Desktop/fado//lib/farsight-0.1-2/librawudp-transmitter.so: > succeeded > > (lt-test-rtp-3:6724): GLib-GObject-WARNING **: > IA__g_object_set_valist: object class `FarsightRTPStream' has no > property named `stun_ip' > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: ensure_local_codecs: media > type is 0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: looking for > /home/g545952/.farsight/gstelements.conf > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: looking for > /home/g545952/Desktop/fado//etc/farsight/gstelements.conf > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdec > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtspsrc > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping rtpdtmfsrc > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec VORBIS with pt > -1, send_pipeline 0x810ad00, receive_pipeline 0x810ac60 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8082630:4:vorbisenc > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093ec0:4:rtpvorbispay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80825a0:4:vorbisdec > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093f50:4:rtpvorbisdepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec MPEG4-GENERIC > with pt -1, send_pipeline 0x810a8c0, receive_pipeline 0x810b270 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80f15a0:4:ffenc_mpeg4 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809cad0:4:rtpmp4gpay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80e81b0:4:ffdec_mpeg4 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809cb60:4:rtpmp4gdepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: skipping codec > audio/unknown, no encoding name specified (pt: 14 clock_rate:90000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec MPA with pt -1, > send_pipeline 0x810b6d0, receive_pipeline 0x810a700 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80f5ad0:6:ffenc_mp2 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f3f0:6:rtpmpapay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8088920:6:mp3parse > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f480:6:rtpmpadepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec PCMU with pt 0, > send_pipeline 0x80e0bd0, receive_pipeline 0x810a650 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093a40:5:mulawenc > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f5a0:5:rtppcmupay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x80939b0:5:mulawdec > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f630:5:rtppcmudepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: adding codec PCMA with pt 8, > send_pipeline 0x810b4b0, receive_pipeline 0x812f540 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093920:5:alawenc > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f510:5:rtppcmapay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: pipeline: > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x8093890:5:alawdec > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: 0x809f6c0:5:rtppcmadepay > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Trying to load > /home/g545952/.farsight/gstcodecs.conf > (lt-test-rtp-3:6724): farsight-DEBUG: Unable to read file > /home/g545952/.farsight/gstcodecs.conf: Aucun fichier ou répertoire de > ce type > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Trying to load > /home/g545952/Desktop/fado//etc/farsight/gstcodecs.conf > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio > speex clock:8000 channels:0 could not be matched with an installed > encoder/decoder/payloader/depayloader quatuor > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio > iLBC clock:8000 channels:0 mode=30 could not be matched with an > installed encoder/decoder/payloader/depayloader quatuor > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio AMR > clock:8000 channels:0 could not be matched with an installed > encoder/decoder/payloader/depayloader quatuor > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio GSM > clock:8000 channels:0 could not be matched with an installed > encoder/decoder/payloader/depayloader quatuor > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio > G729 clock:8000 channels:0 could not be matched with an installed > encoder/decoder/payloader/depayloader quatuor > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Prefered codec -1: audio > PCMU clock:16000 channels:0 could not be matched with an installed > encoder/decoder/payloader/depayloader quatuor > ** Message: codec: 8: PCMA/8000 found > ** Message: codec: 96: MPA/90000 found > ** Message: codec: 0: PCMU/8000 found > ** Message: codec: 97: telephone-event/8000 found > > farsight-rtp-Message: Preparing transmitter > farsight-rtp-Message: connect state changed to 1 > ** Message: state_changed: 0x8063000 connecting > > farsight-transmitter-Message: Media type is 0 > (lt-test-rtp-3:6724): farsight-transmitter-DEBUG: The socket was created > (lt-test-rtp-3:6724): farsight-transmitter-DEBUG: bound to port 7078 > (lt-test-rtp-3:6724): farsight-DEBUG: Interface: lo > (lt-test-rtp-3:6724): farsight-DEBUG: IP Address: 127.0.0.1 > (lt-test-rtp-3:6724): farsight-DEBUG: Ignoring loopback interface > (lt-test-rtp-3:6724): farsight-DEBUG: Interface: eth1 > (lt-test-rtp-3:6724): farsight-DEBUG: IP Address: 192.168.1.9 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Called > farsight_rtp_stream_new_native_candidate > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Native candidates found, > adding to list > native_candidates_prepared: preparation-complete: stream=0x8063000 > ** Message: Local transport candidate: L1 1 UDP RTP 192.168.1.9:7078, > pref 1,000000 > farsight-rtp-Message: connect state changed to 2 > ** Message: state_changed: 0x8063000 connected > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > farsight_rtp_stream_build_base_pipeline (2067): creating core RTP pipeline > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: added transmitter_src > 0x813a048 to pipeline 0x813e060 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > farsight_rtp_stream_set_active_codec (1193): called to change codec > from -1 to 8 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec PCMU 8000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec PCMA 8000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec MPA 90000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: remote_codec telephone-event > 8000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 0: audio PCMU > clock:8000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same > PT, lets try it first > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec > negotiation function > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 0: audio > PCMU clock:8000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 8: audio PCMA > clock:8000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same > PT, lets try it first > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec > negotiation function > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 8: audio > PCMA clock:8000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 96: audio MPA > clock:90000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Have local codec in the same > PT, lets try it first > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Using default codec > negotiation function > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Negotiated codec 96: audio > MPA clock:90000 channels:0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Remote codec 97: audio > telephone-event clock:8000 channels:1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, > local: PCMU, remote: telephone-event > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, > local: PCMA, remote: telephone-event > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Encoding names dont match, > local: MPA, remote: telephone-event > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Could not find a valid > intersection... for codec 97: audio telephone-event clock:8000 channels:1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp, > encoding-name=(string)PCMU, clock-rate=(int)8000, media=(string)audio, > payload=(int)0 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp, > encoding-name=(string)PCMA, clock-rate=(int)8000, media=(string)audio, > payload=(int)8 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: caps are: application/x-rtp, > encoding-name=(string)MPA, clock-rate=(int)90000, media=(string)audio, > payload=(int)96 > > farsight-rtp-Message: farsight_rtp_stream_build_send_pipeline (2662): > creating send pipeline with codec 8 > farsight-rtp-Message: No source has been set yet, send pipeline build > for later > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: We are now PLAYING > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Set_state result was 1 > > > 0x8147400 > new_active_candidate_pair: new-native-candidate-pair: stream=0x8063000 > > > ** (lt-test-rtp-3:6724): DEBUG: Sink needs buffer, must wait for > buffering to stop > > (lt-test-rtp-3:6724): GLib-GObject-WARNING **: > IA__g_object_set_valist: object class `GstAudioTestSrc' has no > property named `latency-time' > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > farsight_rtp_stream_set_source (1738): setting src > farsight-rtp-Message: farsight_rtp_stream_build_send_pipeline (2662): > creating send pipeline with codec 8 > 0x8150520 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: creating send codec bin for > id 8, pipeline_factory 0x810b4b0 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: checking if alawenc is in > config list > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: checking if rtppcmapay is in > config list > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: linking alawenc0 and > rtppmcapay0 with caps audio/x-alaw, channels=(int)1, rate=(int)8000 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: linking src 0x8148d80 to > codec bin 0x8155800 with caps NULL > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found audio/telephone-event > with PT 97 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Looking for PCMA or PCMU in > the remote codecs > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMU codec with PT 0 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMA codec with PT 8 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found remote_pcma = 1 - > remote_pcmu = 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found both the dtmfsrc and > rtpdtmfmux installed > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMA found locally? 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMU found locally? 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Looking for PCMA or PCMU in > the remote codecs > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMU codec with PT 0 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found PCMA codec with PT 8 > > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found remote_pcma = 1 - > remote_pcmu = 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: Found both the dtmfsrc and > rtpdtmfmux installed > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMA found locally? 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: PCMU found locally? 1 > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: farsight_rtp_stream_set_sink > (1848): setting sink alsasink > (lt-test-rtp-3:6724): farsight-rtp-DEBUG: > farsight_rtp_stream_set_sink: No codec bin present, setting new sink > for future use > // > > but the problemeis that noting happens after this . can someone please > guide me how to continue ( i mean how to send or receive the stream ) > regards > Souilem Fadi > > ------------------------------------------------------------------------ > Ne gardez plus qu'une seule adresse mail ! 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